• Asia pacific journal of information systems
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• v.34 no.2
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• pp.421-446
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• 2024

Through Information and Communication Technology (ICT), the growth of music streaming platforms has revolutionized music consumption. "Digital Shadow Work" (DSW) refers to unpaid labor in digital spaces, with some prior research on its aspects. However, a comprehensive understanding is hindered by limitations in existing studies such as a lack of universality and dynamic exploration. To address these gaps and enable a comprehensive investigation into the role of DSW within highly versatile digital applications such as digital streaming platforms, this study employs a grounded theory methodology, a qualitative approach well-suited for exploring the intricacies of DSW among users of Chinese music streaming applications over a two-month period, involving longitudinal interviews with nine participants. The study findings elucidate the dynamic nature of DSW perceptions, which fluctuate across different stages of use and change in intensity over time. This study uncovers mixed attitudes towards DSW tasks, and observes a waning enthusiasm for social features over time, prompting some users to consider switching platforms. This study highlights the importance of thoughtful and user-centric feature development to enhance user satisfaction and the understanding of DSW, providing practical design and enhancement implications for music streaming applications.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.28 no.9
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• pp.99-104
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• 2014

This paper presents a unified live-streaming method based on Hypertext Mark-up Language 5(HTML5) for an IP camera which is independent of browsers of clients and is implemented with open-source libraries. Currently, conventional security systems based on analog CCTV cameras are being modified to newer surveillance systems utilizing IP cameras. These cameras offer remote surveillance and monitoring regardless of the device being used at any time, from any location. However, this approach needs live-streaming protocols to be implemented in order to verify real-time video streams and surveillance is possible after installation of separate plug-ins or special software. Recently, live streaming is being conducted through HTML5 using two different standard protocols: HLS and DASH, that works with Apple and Android products respectively. This paper proposes a live-streaming approach that is linked on either of the two protocols which makes the system independent with the browser or OS. The client is possible to monitor real-time video streams without the need of any additional plug-ins. Moreover, by implementing open source libraries, development costs and time were economized. We verified usefulness of the proposed approach through mobile devices and extendability to other various applications of the system.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2021.10a
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• pp.670-672
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• 2021

The video streaming technologies are used in diverse areas including aerospace. UAV (Unmanned Aerial Vehicle) is still being developed so that we are able to use the UAV not only in military areas but also in civil areas. Remotely monitoring and observing a certain area is possible via using video streaming service based on the UAV nowadays. However, UAV is vulnerable to weather conditions, plane's weight, battery capacity and so on, which makes many restrictions of flight time and video streaming services. In order to prolong the video streaming services, we can use a number of UAVs, fly them one by one, and switch video streaming channels. In the paper, we propose the methods how to interchange UAVs' missions and their video streaming channels to provide seamless video stream services.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.493-504
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• 2007

With the advance and proliferation of computer and wireless network technology, it is common to access to network through the wireless network using mobile device. The ratio of using the streaming media out of many applications through the network is increasing not only in the wired network but also in the wireless network. The streaming media is much bigger than other contents and requires more network bandwidth to communicate and more computing resources to process. However the mobile devices have relatively poor computing resource and low network bandwidth. If the streaming media service is provided for mobile devices without any consideration about the network bandwidth and computing power, it is difficult for the client to get the service of high quality. Since especially mobile devices are supported with very limited energy capacity from the battery, the streaming media service should be adjusted to the varying energy state of mobile devices to ensure the complete playback of streaming media. In this paper, we propose a new method to guarantee the complete playback time of the streaming media for the mobile clients by dynamically controlling transmitted frame rate to the client according to the estimated available time of mobile device using battery model reflecting the characteristic of the battery. Since the proposed method controls the number of frames transmitting to the client according to the energy state of the mobile device, the complete playback time is guaranteed to mobile clients.

• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.155-173
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• 2011

Recently, with the digitalization of a broadcasting system and the development of a communication technology, the existing trend of consuming media contents throughout the fixed-displayer and the dedicated channel is being changed. The existing user only could consume media contents under limited time and places because of the fixed-displayer and the dedicated channel. However, with advent of the IP-based terminals and HTTP adaptive streaming which transfer the media sequence according to the user's transmission condition, users become possible to enjoy the media content anytime anywhere. As the result of the alteration of the broadcasting surrounding, users can enjoy the media content while changing his terminals according to their preferences and circumstances. However, in case that users try to consume consecutively the content from last view-point ended in the previous terminal under current HTTP adaptive streaming environment, a user has to remember the last view-point, and then has to apply the view-point to the changed terminal. Thus, for solving this problem, this paper defines "User Profile" for describing the metadata for the chained content consume environment between the terminals. Also, for proving the proposed method, this paper try to demonstrate the proposed method throughout the realization of the system based on Smooth Streaming from Microsoft.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.6
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• pp.1048-1057
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• 2015

In this paper, we propose a fast MAS reallocation method that can avoid reservation conflicts in the DRP process of WiMedia system in real time. WiMedia MAC DRP has been developed aimed QoS support for audio and video transmission that generate traffic periodically. The size variation of the video streaming data causes DRP reservation conflicts frequently, so that DRP could not support QoS of the video streaming service which is delay-sensitive. Fast MAS reallocation method could avoid reservation conflicts in real-time analyzing reservation requests information of the other device. Our proposed method is evaluated with test sequences using the software simulation and implemented in hardware/software with the integrated WiMedia MAC/PHY system.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.328-335
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• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.14 no.6
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• pp.528-535
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• 2004

Ultrasonic Vibrator is designed to achieve the maximum vibration amplitude at 30 kHz by in-cluding a horn (diameter, 40 mm), mechanical vibration amplifier at the top of the ultrasonic vibrator in the system and making the complete system resonate. In addition, it is experimentally visualized by particle imaging velocimetry (PIV) that the acoustic streaming velocity in the gap is at maximum when the gap between the ultrasonic vibrator and stationary plate agrees with the multiples of half-wavelength of the ultrasonic wave. This fact results from the resonance of the sound wave and the theoretical analysis of that is also accomplished and verified by experiment. It is observed that the magnitude of the acoustic streaming dependent upon the gap between the ultrasonic vibrator and stationary plate possibly changes due to the measurement of the average velocity fields of the acoustic streaming induced by the ultrasonic vibration at resonance and non-resonance. There exists extremely small average velocity at non-resonant gaps while the relatively large average velocity exists at resonant gaps compared with non-resonant gaps. It also reveals that there should be larger axial turbulent intensity at the hub region of the vibrator and at the edge of it in the resonant gap where the air streaming velocity is maximized and the flow phenomena is conspicuous than that at the other region. Because the variation of the acoustic streaming velocity at resonant gap is more distinctive than that at non-resonant gap, shear stress increases more in the resonant gap and is also maximized at the center region of the vibrator except the local position of center (r〓0). At the non-resonant gap there should be low values of vorticity distribution, but in contrast to the non-resonant gap, high and negative values of it exist at the center region of the vibrator with respect to the radial direction and in the vicinity of the middle region with respect to the axial direction. Acoustic streaming is noise-free due to the ultrasonic vibration and maintenance-free because of the absence of moving parts. Moreover, the proposed method by acoustic streaming can be utilized to the nano and micro-electro mechanical systems as a driving mechanism in addition to the augmentation of the streaming velocity.